DTS-HD Decoder for PC

DTS bring the movies watching experience to another level since DVD and had been the mainstream ever since. However, for PC user to experience the full potential of DTS, it's very tedious and picky. First, you will need to bitstream the audio in DTS using spdif, then your amplifier and speaker system also need to be DTS certified. Lack either one of it, then the link is totally broken. But, after reading the following news at lav website, I started googling around for more information.

"DTS-HD and the ArcSoft DTS Decoder

If you’re lucky enough to own a copy of ArcSoft’s TotalMedia Theater, there is some good news for you. In combination with the DTS decoder from TMT, LAV Audio is capable of full high quality DTS-HD decoding.

DTS-HD is the most common audio format to be found on Blu-rays, and before LAV Audio, it was quite a chore to get full decoding of this format – not anymore.

With LAV Audio, decoding DTS-HD is as simple as decoding any other audio format. You just need to get the “dtsdecoderdll.dll” from your “Codecs” directory inside the TMT installation, and drop it inside the directory you installed LAV Audio to. Thats all!"

If I understand correctly, this ArcSoft DTS Decoder seem to decode the DTS and DTS-HD on the PC and send to the amplifier as PCM (loseless audio format). Since the audio is sent in PCM format, it mean that the amplifier doesn't need to be DTS certified. Which mean that it improve our audio quality when watching movies on PC. In order to get the ArcSoft DTS Decoder, you need to get a copy of ArcSoft TotalMedia Theatre from http://www.arcsoft.com/totalmedia-theatre/

Technical Results
So, after getting myself a copy of ArcSoft TotalMedia Theatre, I copy the "dtsdecoderdll.dll" to the same folder where "LAVAudio.ax" located. For this, you need to be using LAV Audio decoder as the main audio decoder which is by default in the latest K-Lite or CCCP (i personally recommend CCCP over K-Lite). Or you can try my personal prefer Media Player setup here:
http://casualnfunentertainment.blogspot.com/2013/06/media-player-setup-for-pc.html

We will first look at some screenshots to see the technical differences before and after using "dtsdecoderdll.dll":

Before using "dtsdecoderdll.dll":

After using "dtsdecoderdll.dll":

From these screenshots, we can notice that before using "dtsdecoderdll.dll", it can only decode the audio up to 48kHz. However, after using the "dtsdecoderdll.dll", it is able to decode the audio as how it is originally recorded (which is 192kHz for this violin concert but for movies it is usually just 24bit/48kHz).

Listening Results on DTS-HD MA
That's enough for the technical results. Listening test is still the most important. I listened to few different audios, live recordings and movies that have DTS-HD MA audio. When comparing to LAV Audio, the differences is like day and night. The audio produced by "dtsdecoderdll.dll" just has so much more details and realistic feel to it. The effect just have more power in them. When comparing to Ffdshow Audio, the differences is not so significant as comparing to LAV Audio. But, the audio produced by "dtsdecoderdll.dll" still has more details and realistic feel in it. The effect also have more power in them.

Listening Results on DTS
I think many users will like to know whether the "dtsdecoderdll.dll" improve normal DTS audio quality also as most of the mkv movies has their audio strip down to 1.5Mbps DTS only. I tried few mkv and old DVD iso that has DTS audio. And the answer is yes, the improvement is the same as on DTS-HD MA. I heard day and night different when comparing to LAV Audio and heard some improvement when comparing to Ffdshow Audio.

Verdict
- The "dtsdecoderdll.dll" is the best DTS decoder on PC that I had tried so far.
- The audio produced by "dtsdecoderdll.dll" has more details and realistic feel.
- The "dtsdecoderdll.dll" is even able to decode the audio above 48kHz (up to 192kHz in the test, maybe I don't have higher quality audio file)

Jplay Just Turn My PC Into A Hifi Music Player

When I first came through the name of Jplay, I was thinking it is most likely just another gimmick or hoax that intend to sell more than it can do. Will Jplay, a software music player that also used kernel streaming improve the sound of my PC that already using kernel streaming? That is the doubt I have in my mind at first.

So I went and google about Jplay. Surprisingly, Jplay has very good and positive feedbacks and reviews. The 6moons.com even gave Jplay a "Blue Moon Award" with the praise "Innovation in sound quality & musicality for PC / Windows music player software". There are also feedback from users stating that they even heard improvement on sound quality from their integrated / onboard sound card. These feedbacks and reviews seem very convincing.

Free Trial Version
This is definitely the best part of Jplay. They provide free trial version to be downloaded and tested with all the features enabled. Just that there will be moments of silent that happen occasionally when you are listening to the music. So you are free to judge whether that Jplay improve your sound quality and whether it worth the money. Here is the link for the free trial version of Jplay: http://jplay.eu/download/

Jplay Sound Quality
At the first play after installed the trial version, I am totally sold by the improved sound quality. The sound just seem to be so relaxing and airy feel that I never heard on my DAC before. That is just the default settings on Jplay. After messing around with the setting of Jplay, I think Xtream setting with Direct Link seem to be the best while my sister prefer the River setting. Well, it actually depend on personal taste. From other users' feedback, it seem that the ULTRAstream setting that only support windows 8 is even better. But I will just leave it at Xtream setting for now since I don't have any plan to upgrade to win 8 anytime soon.

Verdict
- Jplay really does improve your sound from the PC.
- However, Jplay is also very demanding on PC requirement.
- With the price tag of 99 Euro Dollar. Jplay is not for everyone. It is means for those in pursuit of higher quality sound that owned some system that cost at least RM 5k or above.
- Try it out yourself with the trial version here: http://jplay.eu/download/

Does USB Cable Affect Sound Quality

Does a USB cable really affect sound quality? I think this question had already been asked many times at many forums and always ended up with long debates. And with the concept of spend only 10% of the budget into accessories, I never really got the chance to verify this issue myself as I do not own such an expensive DAC. However, I finally manage to get myself a RM2.4k Ciunas DAC last month. Therefore, I am eager to try out whether a USB cable really do affect the sound quality or not. The DAC used a A to B USB cable which is commonly found on printer. Hence, I gather around all the cables I am able to find in my house and get ready for testing. Just a side note, this DAC is power up by another USB charger cable which is included. Therefore, the A to B USB cable should only be affecting the sound part and not the electrical part.

USB Cable Do Make A Difference
After testing around with those free A to B USB cables that I found, I am able to hear significant differences when switching the cables. After being convinced on this, I started to look for better audio USB cable. But, it's quite disappointing and frustrated as the audio USB cable market is not very popular and flooded as analogue interconnect. After looking for a while, I manage to purchase a 2m DH Labs Silversonic USB cable and a 1.2m Furutech GT2 USB cable.

Listening Test
I run in these 2 cables each for 100 hours before beginning the test. The DH Labs and Furutech cables are totally opposite characteristic of each other. The DH Labs cable have very good control over the bass but the vocal seem to be a little dark or cold. Meanwhile, the Furutech cable has very good dynamic expression in the vocal but the bass seem to getting bloomy times to times. Then, I switch back to the free cable that I think was the best before. I straight away notice that the bass, treble and vocal totally lack of focus and power when comparing to those 2 cables. The characteristic of this free cable seem to be more of the same as the Furutech cable. Below are the pics of these 3 cables.

Free cable:

DH Labs Silversonic:

Furutech GT2:

Recordings
As usual, I recorded the music using my handphone. But, this time I recorded 2 tracks for each cable; 1 is male vocal while another is female vocal to show the characteristic of the cables. The setup is Ciunas DAC, Marantz pm ki pearl and Kef ls50.

Free cable:


DH Labs Silversonic:


Furutech GT2:

These recordings just serve as references and may not present the live listening condition.

Verdict
- USB cable DO affect the sound quality
- following the concept that only 10% of the budget should allocate for accessories, the USB cable price should be 10% of the DAC price or at most 20%
- the sound of those free cables aren't that bad
- each cable has their own characteristic. There is no best cable and expensive cable may not always be better. It is very important to match the whole setup and your personal listening taste.
- we don't have the return policy that available at some countries and this made us hard to find cables that match our setup and listening taste. Not only USB cable, I am including speaker cable, analogue interconnect, power cable and etc.
- I am thinking maybe the rapid growth and improvement on the DAC market bring forth to this long debate. If you think that USB cable do not make any differences with your old and maybe expensive DAC, just try to borrow a latest model DAC that cost at least RM3k to test it out. Maybe the result will be different.

Below is some theories that I found on a local forum that describes why USB cable affect sound quality even it is passing "0" and "1". It is not proven and lengthy. Read it if you are interested or just skip it.

Basically the USB connection is just an electrical connection. It transmits an electric signal, measured in terms of voltage and current, though voltage is the parameter we are mainly interested in. So the USB cable is susceptible to all those things which affects the standard electrical signal transfer. The main advantage of a digital signal is again that we limit ourselves to two voltage levels, e.g. 0V and 1V. But in real life, the signal can vary anywhere between 0V and 1V. I already explained the issue of setting the threshold for defining digital-0 and digital-1.

Then there is the issue of synchronization. This is the issue of EXACTLY WHEN the signal changes. E.g. if I want to send a digital stream - 001100. Then I modulate the signal in the USB cable to 0V, 0V, 1V, 1V, 0V, 0V. OK then, EXACTLY WHEN do I do that in terms of TIME. And for the purpose of transmission, EXACTLY WHEN do the receiver (e.g. the DAC) 'sample' or measure the voltage in the USB cable to determine what I am sending ? This is the issue of synchronisation.

Now the question of EXACTLY WHEN is the fundamental and most problematic one. TIME is a CONTINOUS entity in so far as our science is concerned, right down to quantum physics. We have a limited ability to deal with it in terms of our maths etc but in quantum physics, TIME and ENERGY levels are continous entities. OK, I will not dwell into this anymore, you can google the net if you want to read more. 

A USB connection works with a certain 'clock rate'. What this means is the 'sampling rate' by which the sender sends and the receiver must measure the signal. For a USB connection this can range from 40 to 60MHz for e.g., i.e. the sender -and receiver - works with a sample, or signal change, of between 40 to 60 MILLION times a second. Put another way, the sender sends a signal - changes the voltage in the connection - once every 0.000000025 to 0.0000000167.... seconds. 

NOTE one anomaly here - the 2nd value above is an approximate. I.e. a 60MHz works out to an exact value of ?.... WE CANNOT express it accurately, it is 0.000000016666666........ INFINITELY. So if we want to work with it, we HAVE to make a decision to adopt a fixed DEGREE of accuray, say 10 decimal places. Therefore 60MHz is 0.0000000167 seconds APPROXIMATELY. This is another issue which forces it way in all scientific and technology discussion. The LIMITATION of our scientific foundation, in this instance, the limitation of the most basic of our science - our NUMBERING system. Again, I won't talk about it anymore and you can google if you want to know more.

So basically the important issue about synchronisation is simply that the sender and receiver MUST work at EXACTLY the same time, or for practical purpose, given the limitations of our technology, at a level of accuracy high enough to ensure there is no error. Meaning that if the receiver tries to sample the signal at the wrong time, then it will be reading a wrong value. And as you can see, it needs to be very accurate. 

So, imagine an electrical signal changing once every 0.0000000167 seconds. Now imagine how sensitive it can be to interference, to cable material quality, to sender/receiver circuitry, and all the other things we know can affect an electrical signal. 

This is why it is NEVER just a series of 1's and 0's. It NEVER EVER was. 

The USB connection has NOT and WILL NOT be adapted by the commercial or military/space environment for their critical communications. I.e. equipments in a massive mainframe computer or a massive server farm are NOT connected using USB. These environments are those which demands the max in accuracy and reliability. So think about what is says about this interface standard. Now translate it to what you need to do to ensure good accuracy and reliability.

Is Power Conditioner Or PLC Another improvement to the Sound

After setting up the Edifier C3 with some speaker isolation, I try to find more approaches that able to improve the sound. So after googling and asking around, I plan to improve the sound of Edifier C3 by conditioning my power source. Therefore, I started to search for a budget power conditioner aka PLC. In my search for a power conditioner, I came to summarize it to 2 main categories. 1 is the passive type while another is the active type. We can go deeper into the categories and theories but that is not casual anymore XD.

From my understanding, the passive type power conditioner:
- tend to condition the power source without affecting the original sound as much as possible
- able to reduce the background noises in a song
- most likely able to protect your gadgets from power surge or lightning (it also depend on the lightning power)

The active type power conditioner:
- the sound is being pass through filter that may affect the sound
- tend to be able to reduce the background noises in a song better than passive type at the same price but at the same time the song may lost some details in the process
- mostly likely to have more extra features than passive type at the same price

And there seem to be something to be taken note before choosing to use power conditioner or not:
- power conditioners will take away some of the dynamic in the sound although these became less and less significant with more expensive model
- this problem seem to be non noticeable only on models that priced around RM 10k or more
- these lost of dynamic in sound is more significant if your amplifier draw more power

Soundstage SAVR-2000 (Active Type Power Conditioner)
After knowing these facts, I look back at my setup, Edifier C3 and a budget desktop with built in graphic. These seem to be low power consumption. Therefore, I went and purchased a budget active type power conditioner for the extra feature which is Automatic Voltage Regulator (AVR). I bought it at RM 999 and it is branded as Soundstage (a local company) with the model name SAVR-2000.

Testing
I plug my desktop and Edifier C3 straight to the SAVR-2000 and power it up. There is only 2 settings for me to play around which are the "Delay" and "Precision". I can set the Delay to either 3 minutes or 6 seconds. Setting the Delay to 6 seconds resulted in a lot more background noise when comparing to 3 minutes. The Precision setting don't seem to have much noticeable different.

Below is the picture of the power conditioner when powering my desktop and C3. The current (A) column always show 0.0 and sometimes show 0.1 which mean that this setup is very low power consumption. Change the desktop to my gaming dekstop, "pop" the current (A) column straight away jump to 2.2 at idle @@. Gaming rig sure are power hunger -.-'''

Results
So I started the test with the Delay set at 3 minutes and Precision set at High. The setup is the Edifier C3 with speaker isolation connected to UD100. The music player is Foobar2000 with kernel streaming.

After having the power conditioner, the background noises are reduced a lot. The bass seem to be better controlled without getting so bloomy. This is maybe due to the noticeable reduce in dynamic of the song. The vocal seem to be color as this is an active type power conditioner. It depend on personal tastes whether the coloration is good or bad. For me, the coloration is acceptable. Below is the recordings from my phone with the song played at 60% volume and bass at -3.

Without power conditioner:

With power conditioner:

These recording just serve as references and may not fully represent live listening condition.

Conclusion
The price for a budget power conditioner seem to be a little bit steep for casual users. However, it does have improvements such as:
- lower background noise
- better controlled bass
with some flaws as well:
- loss of dynamic
- coloration of vocal

Therefore, if you prefer quiet listening experience, it is worth to get a power conditioner. If not, I personally think that try getting a better speaker + amplifier + DAC that total cost over RM5k or RM10k first before getting a power conditioner.

Better Audio from Videos by Using ReClock

After setting up the music player with kernel streaming to get better audio quality, I tried few programs which are Virtual Audio Cable, AsioRenderer, Asio4all, and ReClock that been said able to improve audio quality from a video. After testing all 4, I personally think that ReClock is the best as it is easy to setup and working well. You can download Reclock for free from:

http://forum.slysoft.com/showthread.php?s=2165e84dd8a021dff42f972579f4e877&t=56678

1) After downloaded ReClock, just do a Full Installation with everything checked.

2) Then open ReClock. At the "Audio settings" tab, UNTICK everything.

    A) Set PCM to "WASAPI Exclusive".

    B) Pre-buffer Duration, leave it as "500". 

         OPTIONAL: You can test around and see which setting suit your personal taste ranging 25 to 1000.

    C) For Format set it to "24 bit int padded to 32" while for Sampling rate, it can be leave as "Same as input (default)" for normal users or UD100. 

         OPTIONAL: If your DAC is capable of better setting then set Sampling rate to your DAC max setting such as 96kHz, 192kHz or 384kHz.

    D) Set Quality to "Best Sinc Interpolation".

3) Go to "Video settings" tab

UNTICK "Use built-in estimator (recommended)" on both side as these settings may chuck up the videos after few seconds of opening.

4) Go to "Advanced settings" tab and UNTICK everything except (B) "Force ReClock to be loaded instead of default renderers (recommended)".
 
    OPTIONAL: TICK (A)"Give high CPU priority to player", this setting will let your player have higher priority over other software and ensure less lagging and chuck up.

5) Click "OK" and ReClock is ready to use. So everytime you open a supported media player such as Media Player Classic, Zoom Player, PowerDVD and etc, a box will pop up and ask whether to load ReClock or not. You can either choose to always load or never load. These setting can be change in (C).

Verdict
As usual, some may prefer the newly setup sound while some may not. If you do not prefer the newly setup sound, just install Reclock. Personally, i think it does improve the sound from a video by:
- better seperation of vocal and instruments
- less noise

Cheap Speaker Isolation

In the audio world, everything also affect the sound quality either by a large or small margin. So after getting a budget DAC and kernel streaming, this time I tried some cheap speaker isolation. By googling "speaker isolation", I get so many results of spike and feet in different materials, sizes and shapes:

So what is speaker isolation? To make it short, speaker isolation is to isolate your speaker from touching unwanted surface. This is to reduce the chances of the sound produce by the speaker getting influence by unwanted external element. This is being done by putting some isolation product in between your speaker and the surface such as table or floor. Isolation product come in many different materials, sizes and shapes. It can be spikes as shown in the pictures above, it can also be a block of wood or marble plates. You can also setup few types of isolation products together as long as it suit your listening taste. So, it's up to your personal preferences and availability of isolation product in your area. 

Testing

So with a limited budget (RM10), I tried 2 different types of speaker isolation (i will refer as isolation A and B) on the following setup:

Speaker: Edifier C3

DAC: Stoner Acoustics UD100

Music Player: Foobar2000 with Kernel Streaming

The isolation is being carried out on the 2 satellites and the subwoofer.

Result

I heard significant differences between the 3 setup (without any isolation, isolation A, isolation B). Both speaker isolation setups have significant improvement on the overall sound quality as the bass, treble and vocal became more solid and have more power. Sound separation also seem to be better with speaker isolation. 

As to choose between the speaker isolation setup, all I can say is different material and size do give different sound. For shapes, I am not really sure as the 2 types of isolation I tested are of the same shape. You will also need to consider your speaker size and weight when choosing the speaker isolation setup. And it's all up to personal preferences and I can't really recommend which is better. You have to test it yourself.

I decided to record the same song with and without speaker isolation. It is recorded using my handphone. The song is played at 70% volume (i turn the volume up as the recording seem a little too soft) with bass -3 (i turn down the bass as i seem to be a little too heavy).

Without speaker isolation:

With speaker isolation:

The recordings just serve as references as speaker isolation do improve sound quality and may not fully represent live condition.

Conclusion

- Cheap speaker isolation (budget less than RM10) do improve sound quality.

- Different size and material of speaker isolation product have different improvement.

- Speaker size and weight also need to be consider when choosing speaker isolation product.

- Make sure the speaker is stable after isolation and won't be easy to knock over.

Music Playback with Kernel Streaming Just Make the DAC even Better ^^

After getting a DAC such as the recommended Stoner Acoustics UD100 in my previous post or other DAC,  its time to focus back on the music playback software on the pc. There are many positive feedback about setting up the audio player with "Kernel streaming", "ASIO" or "WASAPI-exclusive" can improve audio quality. Personally, i prefer "kernel streaming" over the other settings and will only focus on it in this post.

For the music player part, i personally prefer "ttplayer" (千千静听) and "foobar2000". Both players are also free to download and use. Ttplayer seem to focus more on the vocal part of the song while the instrument part seem a little bit less focus. Besides that, this player also seem to color the vocal a bit to be more sweet to listen to. I used this player when I wanted to listen to some sweet vocal especially during Chinese songs. This player is definitely not suitable for those that wanted the natural or original sound. Foobar2000 is a more natural and balance sounding player. It's the main player on my pc.

Setting Up Kernel Streaming with Ttplayer (千千静听)
Ttplayer already have kernel streaming included. You just have to set it in the settings as long as the DAC support it.

1) Choose setting (千千选项) in Ttplayer.

2) Set it as shown in the picture below. Click save and it's well to go.

Setting Up Kernel Streaming with Foobar2000
Foobar doesn't include kernel streaming when installed.

1) Download the Kernel Streaming for Foobar from: http://www.foobar2000.org/components/view/foo_out_ks
2) Then open Foobar2000 player.
3) Choose "Library" > "Configure".

4) Choose "Components" > "Install".

5) Choose the downloaded zip file without extracting it.
6) Click "OK" and restart Foobar.
7) Go "Library" > "Configure" again.
8) Choose "Playback" > "Output", then choose "KS: (DAC name)".

Verdict
Some may prefer the newly setup sound while some may not. It's totally up to your personal taste. If you don't like the sound of Kernel Streaming, you can just change it back to the old setting. But personally, i think kernel streaming really does improve the sound quality by:
- better separation of vocal and also instruments
- clearer and more power sounding
- less noise
But it is not without flaw. The flaw of kernel streaming is only 1 software can use the DAC at the same time. Sometimes you will found that the flash player of internet browser is hogging the DAC and you are unable to play any song on the player.